- 21 Mar, 2013 1 commit
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David Henningsson authored
Headset mic jacks, i e TRRS style jacks with Headphone Left, Headphone Right, Mic and GND signals, are becoming increasingly common and are now being shipped by several manufacturers. Unfortunately, the HDA specification does not give us any hint of whether a Mic pin belongs to such a jack or not, but it would still be helpful for the user to know (especially if there is one TRS Mic jack and one TRRS headset jack). This new fixup causes the first (non-dock, non-internal) mic to be a headset mic jack. The algorithm can be later refined if needed. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 18 Mar, 2013 12 commits
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Takashi Iwai authored
The new HP desktop machines have Realtek codecs and their LEDs are controlled via GPIO as for many laptop models. Add similar hooks as well as in patch_sigmatel.c for controlling LEDs. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
The verb to set up the digital beep via AC_VERB_SET_DIGI_CONVERT_2 should be executed at resume as well. Use the cached write for it being performed automatically at resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
While playing the digital beep tone, the codec shouldn't be turned off. This patch adds proper snd_hda_power_up()/down() calls at each time when the beep is played or off. Also, this fixes automatically an unnecessary codec power-up at detaching the beep device when the beep isn't being played. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Instead of calling snd_hda_attach_beep_device() and snd_hda_detach_beep_device() in each codec driver, move them to the generic parser. The codec driver just needs to set spec->beep_nid for activating the digital beep. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Back-merged for refactoring beep stuff.
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Takashi Iwai authored
The argument passed to snd_hda_attach_beep_device() is a widget NID while spec->beep_amp holds the composed value for amp controls. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
This leaks the beep input device after module unload, which leads to Oops. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=55321Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Daniel Mack authored
"Playback Design" products need a 50ms delay after setting the USB interface. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Daniel Mack authored
UAC2 compliant audio devices may announce the capability to transport raw audio data on their endpoints. Catch this and handle it as 'special' stream on the ALSA side. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Daniel Mack authored
This field may use up to 32 bits, so it should be handled as unsigned int. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Andreas Koch <andreas@akdesigninc.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mark Hills authored
The maxpacksize field is given in some quirks, but it gets ignored (in favour of wMaxPacketSize from the first endpoint.) This patch favours the one in the quirk. Digidesign Mbox and Mbox 2 are the only affected quirks and the devices are assumed to be working without this patch. So for safety against the values in the quirk being incorrect, remove them. The datainterval is also ignored but there are not currently any quirks which choose to override this. Cc: Damien Zammit <damien@zamaudio.com> Cc: Chris J Arges <christopherarges@gmail.com> Signed-off-by: Mark Hills <mark@xwax.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Mark Hills authored
The hardware also has a PCM capture device which is not implemented in this patch. It may be possible to generalise this to Saffire 6 USB support and some of the other Focusrite interfaces, but as I don't have access to these devices we should wait until capture support is working first. Capture support is not implemented because the code assumes the endpoint to have its own interface (instead, it shares the interface with playback) and some thought will be needed to lift this limitation. Signed-off-by: Mark Hills <mark@xwax.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 17 Mar, 2013 1 commit
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Masanari Iida authored
Correct spelling typos in Documentation/sound/alsa Signed-off-by: Masanari Iida <standby24x7@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 15 Mar, 2013 5 commits
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Takashi Iwai authored
During the transition to the generic parser, the hook to the codec specific automute function was forgotten. This resulted in the silent output on some MacBooks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dan Carpenter authored
"chn" here is a number between 0 and 255, but ->chn_info[] only has 16 elements so there is a potential write beyond the end of the array. If the seq_mode isn't SEQ_2 then we let the individual drivers (either opl3.c or midi_synth.c) handle it. Those functions all do a bounds check on "chn" so I haven't changed anything here. The opl3.c driver has up to 18 channels and not 16. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dylan Reid authored
spec->dsp_state is initialized to DSP_DOWNLOAD_INIT, no need to reset and check it in ca0132_download_dsp(). Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dylan Reid authored
Instead of using the dspload_is_loaded() function, check the dsp_state that is kept in the spec. The dspload_is_loaded() function returns true if the DSP transfer was never started. This false-positive leads to multiple second delays when ca0132_setup_efaults() times out on each write. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dylan Reid authored
If dspload_image() fails, it was ignored and dspload_wait_loaded() was still called. dsp_loaded should never be set to true in this case, skip it. The check in dspload_wait_loaded() return true if the DSP is loaded or if it never started. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 14 Mar, 2013 1 commit
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David Henningsson authored
If there are no internal speakers, we should not turn the eapd switch off, because it might be necessary to keep high for Headphone. BugLink: https://bugs.launchpad.net/bugs/1155016Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 13 Mar, 2013 5 commits
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Takashi Iwai authored
So far, the driver doesn't power down the widget at going down to D3 when the widget node has an EAPD capability and EAPD is actually set on all codecs unless codec->power_filter is set explicitly. This caused a problem on some Conexant codecs, leading to click noises, and we set it as NULL there. But it is very unlikely that the problem hits only these codecs. Looking back at the development history, this workaround for EAPD was introduced just for some laptops with STAC9200 codec, then we applied it blindly for all. Now, since it's revealed to have an ill effect, we should disable this workaround per default and apply only for the known requiring systems. The EAPD workaround is implemented now as snd_hda_codec_eapd_power_filter(), and this has to be set explicitly by the codec driver when needed. As of now, only patch_stac9200() sets this one. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Use the dynamic array allocations for pins, converters and PCM arrays instead of the fixed size arrays. The modern HDMI codecs get more and more pins, and we don't know the sensitive limit. Most of the patch are spent for the straight conversions from the fixed array access to snd_array helpers. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
Make sure that the allocated buffer for reading the proc file won't expose the uncleared kernel memory. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 12 Mar, 2013 4 commits
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Takashi Iwai authored
In the connection list expansion in hda_codec.c and hda_proc.c, the value returned from snd_hda_get_num_raw_conns() is used as the array size to store the connection list. However, the function returns simply a raw value of the AC_PAR_CONNLIST_LEN parameter, and the widget list with ranges isn't considered there. Thus it may return a smaller size than the actual list, which results in -ENOSPC in snd_hda_get_raw_conections(). This patch fixes the bug by parsing the connection list correctly also for snd_hda_get_num_raw_conns(). Reported-and-tested-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Clemens Ladisch authored
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the interface associations come out wrong, which results in the driver erroring out with the message "Audio class v2 interfaces need an interface association". Work around this by searching for the interface association descriptor also in some other place where it might have ended up. Reported-and-tested-by: Dave Helstroom <helstroom@google.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Wei Yongjun authored
The dereference should be moved below the NULL test. Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Yacine Belkadi authored
script/kernel-doc reports the following type of warnings (when run in verbose mode): Warning(sound/core/init.c:152): No description found for return value of 'snd_card_create' To fix that: - add missing descriptions of function return values - use "Return:" sections to describe those return values Along the way: - complete some descriptions - fix some typos Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 11 Mar, 2013 10 commits
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Adrian Knoth authored
Expose the newly added TCO LTC and sync check functions to userspace. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
This patch adds new ALSA controls to query the LTC state from userspace. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
This patch prepares snd_hdspm_get_sync_check() to also check the TCO sync state. The added feature will be exposed to the user in a later commit. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
Considerably shorten the code by using a macro. Though this won't lower the binary size, it makes the source more readable. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
For 96kHz, MADI allows to multiplex the samples (SMUX) or to use a dedicated 96K mode. The RME cards default to 96K mode, but since not all external MADI equipment supports this, provide a switch to users that changes the on-wire protocol to SMUX. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
When using the additional Time Code Option module in slave mode or the SYNC-In wordclock connector, the sample rate needs to be returned by hdspm_external_sample_rate(). Since this sample rate may contain any value with 1Hz granularity, we need to round it to a common rate as done by the OSX driver. [Fixed missing function declarations by tiwai] Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Adrian Knoth authored
This commit introduces hdspm_get_pll_freq() to avoid code duplication. Reading the sample rate from the DDS register will be required by upcoming code. Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
When a headphone pin is set up as a shared hp/mic pin, we rather want to keep it as a headphone primarily as default, but the driver overrides it always as a mic pin, just because the input controls are created after outputs. Add a check of pin NID and skip the re-initialization of pinctl for such a shared hp/mic pin. Reported-by: Jonathan Woithe <jwoithe@just42.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Takashi Iwai authored
snd_seq_timer_open() didn't catch the whole error path but let through if the timer id is a slave. This may lead to Oops by accessing the uninitialized pointer. BUG: unable to handle kernel NULL pointer dereference at 00000000000002ae IP: [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 PGD 785cd067 PUD 76964067 PMD 0 Oops: 0002 [#4] SMP CPU 0 Pid: 4288, comm: trinity-child7 Tainted: G D W 3.9.0-rc1+ #100 Bochs Bochs RIP: 0010:[<ffffffff819b3477>] [<ffffffff819b3477>] snd_seq_timer_open+0xe7/0x130 RSP: 0018:ffff88006ece7d38 EFLAGS: 00010246 RAX: 0000000000000286 RBX: ffff88007851b400 RCX: 0000000000000000 RDX: 000000000000ffff RSI: ffff88006ece7d58 RDI: ffff88006ece7d38 RBP: ffff88006ece7d98 R08: 000000000000000a R09: 000000000000fffe R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: ffff8800792c5400 R14: 0000000000e8f000 R15: 0000000000000007 FS: 00007f7aaa650700(0000) GS:ffff88007f800000(0000) GS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00000000000002ae CR3: 000000006efec000 CR4: 00000000000006f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000ffff0ff0 DR7: 0000000000000400 Process trinity-child7 (pid: 4288, threadinfo ffff88006ece6000, task ffff880076a8a290) Stack: 0000000000000286 ffffffff828f2be0 ffff88006ece7d58 ffffffff810f354d 65636e6575716573 2065756575712072 ffff8800792c0030 0000000000000000 ffff88006ece7d98 ffff8800792c5400 ffff88007851b400 ffff8800792c5520 Call Trace: [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff819b17e9>] snd_seq_queue_timer_open+0x29/0x70 [<ffffffff819ae01a>] snd_seq_ioctl_set_queue_timer+0xda/0x120 [<ffffffff819acb9b>] snd_seq_do_ioctl+0x9b/0xd0 [<ffffffff819acbe0>] snd_seq_ioctl+0x10/0x20 [<ffffffff811b9542>] do_vfs_ioctl+0x522/0x570 [<ffffffff8130a4b3>] ? file_has_perm+0x83/0xa0 [<ffffffff810f354d>] ? trace_hardirqs_on+0xd/0x10 [<ffffffff811b95ed>] sys_ioctl+0x5d/0xa0 [<ffffffff813663fe>] ? trace_hardirqs_on_thunk+0x3a/0x3f [<ffffffff81faed69>] system_call_fastpath+0x16/0x1b Reported-and-tested-by: Tommi Rantala <tt.rantala@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Christine Spang authored
Having snd_BUG_ON() only evaluate its conditional when CONFIG_SND_DEBUG is set leads to frequent bugs, since other similar macros in the kernel have different behavior. Let's make snd_BUG_ON() act like those macros so it will stop being accidentally misused. Signed-off-by: Christine Spang <christine.spang@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- 07 Mar, 2013 1 commit
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Takashi Iwai authored
When the headphone mic jack enum control is created (via explicitly specification by user), it doesn't make much sense to change the I/O direction dynamically per capture source change, since the I/O direction is rather controlled over the enum ctl. This also reduces the implicit dependency between the capture source and the hp mic jack enum ctls, which might confuse a program accessing the whole control elements at once like alsactl. In addition, this patch introduces update_hp_automute_hook() function to call the proper hook function. It's just to remove the open codes in multiple places in hda_generic.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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